АТС - Asterisk 1.8.7.2 built by root @ evil on a amd64 running FreeBSD
Учетка - 295039.
sip.conf
- Код: Выделить всё
[general]
register => 295039:***@sip.comtube.ru/295039
[5555]
type=friend
username=5555
secret=XXX
callerid="IT" <5555>
host=dynamic
nat=yes
canreinvite=no
disallow=all
allow=ulaw
allow=alaw
dtmfmode=auto
;callgroup=1,2-4
pickupgroup=1
context=office
transport=tcp
transport=udp
[comtube-out]
type=peer
insecure=very
;transport=tcp
transport=udp
hassip=yes
username=295039
secret=***
host=sip.comtube.ru
fromuser=295039
fromdomain=sip.comtube.ru
dtmfmode=auto
canreinvite=no
qualify=yes
disallow=all
allow=ulaw
allow=alaw
allow=g729
nat=yes
context=from-comtube
dialplan
- Код: Выделить всё
[to-comtube]
exten => _7./XXXX,1,Dial(SIP/comtube-out/${EXTEN},120,rt)
exten => _7.,n,Hangup
Сложилась интересная ситуация. При попытке позвонить получаем следующее:
- Код: Выделить всё
<--- SIP read from UDP:62.117.120.98:5060 --->
SIP/2.0 100 Giving a try
Via: SIP/2.0/UDP 193.169.37.103:5060;received=193.169.37.103;branch=z9hG4bK7a994bf1;rport=5060
From: "IT" <sip:295039@sip.comtube.ru>;tag=as33b2d629
To: <sip:79242245055@sip.comtube.ru>
Call-ID: 6357fbec6699dbc17afe09692e3144ea@sip.comtube.ru
CSeq: 104 INVITE
Server: Comtube SIP Proxy
Content-Length: 0
<--- SIP read from UDP:62.117.120.98:5060 --->
SIP/2.0 403 Forbidden
From:sip:295039@sip.comtube.ru;tag=as33b2d629
To:<sip:79242245055@sip.comtube.ru>;tag=2EBB3030353039320011CA41
Call-ID:6357fbec6699dbc17afe09692e3144ea@sip.comtube.ru
CSeq:104 INVITE
Server:TB005092/2.1
Via:SIP/2.0/UDP 193.169.37.103:5060;received=193.169.37.103;branch=z9hG4bK7a994bf1;rport=5060
Content-Length:0
<------------->
--- (8 headers 0 lines) ---
Transmitting (NAT) to 62.117.120.98:5060:
ACK sip:79242245055@sip.comtube.ru SIP/2.0
Via: SIP/2.0/UDP 193.169.37.103:5060;branch=z9hG4bK7a994bf1;rport
Max-Forwards: 70
From: "IT" <sip:295039@sip.comtube.ru>;tag=as33b2d629
To: <sip:79242245055@sip.comtube.ru>;tag=2EBB3030353039320011CA41
Contact: <sip:295039@193.169.37.103:5060>
Call-ID: 6357fbec6699dbc17afe09692e3144ea@sip.comtube.ru
CSeq: 104 ACK
User-Agent: Asterisk PBX 1.8.7.2
Content-Length: 0
[Jul 4 16:17:24] WARNING[8452]: chan_sip.c:19680 handle_response_invite: Received response: "Forbidden" from '"IT" <sip:295039@sip.comtube.ru>;tag=as33b2d629'
-- SIP/comtube-out-00000008 is circuit-busy
== Everyone is busy/congested at this time (1:0/1/0)
-- Auto fallthrough, channel 'SIP/5555-00000007' status is 'CONGESTION'
<--- Reliably Transmitting (NAT) to 192.168.1.82:5060 --->
SIP/2.0 503 Service Unavailable
Via: SIP/2.0/UDP 192.168.1.82:5060;branch=z9hG4bK4cbc51b58a63cbeef523fdca47033b8;received=192.168.1.82;rport=5060
From: "OIT" <sip:5555@192.168.1.1>;tag=107713114
To: <sip:79242245055@192.168.1.1;user=phone>;tag=as0c68134a
Call-ID: 239705914@192_168_1_82
CSeq: 3 INVITE
Server: Asterisk PBX 1.8.7.2
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
-Asterisk-HangupCause: Call Rejected
X-Asterisk-HangupCauseCode: 21
Content-Length: 0
<------------>
Really destroying SIP dialog '3832e75741d8f5245078e4d12248bd2e@sip.comtube.ru' Method: INVITE
<--- SIP read from UDP:192.168.1.82:5060 --->
ACK sip:79242245055@192.168.1.1;user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.1.82:5060;branch=z9hG4bK4cbc51b58a63cbeef523fdca47033b8;rport
From: "OIT" <sip:5555@192.168.1.1>;tag=107713114
To: <sip:79242245055@192.168.1.1;user=phone>;tag=as0c68134a
Call-ID: 239705914@192_168_1_82
CSeq: 3 ACK
Contact: <sip:5555@192.168.1.82:5060>
Authorization: Digest username="5555", realm="asterisk", algorithm=MD5, uri="sip:79242245055@192.168.1.1;user=phone", nonce="4cc04188", response="f10397c2862457a20ed857b35f630a0d"
Max-Forwards: 70
User-Agent: C595 IP/42.049.00.000.000
Content-Length: 0
<------------->
--- (11 headers 0 lines) ---
Really destroying SIP dialog '239705914@192_168_1_82' Method: ACK
<--- SIP read from UDP:62.117.120.98:5060 --->
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 193.169.37.103:5060;received=193.169.37.103;branch=z9hG4bK74020623;rport=5060
From: "IT" <sip:295039@sip.comtube.ru>;tag=as4785c625
To: <sip:79242245055@sip.comtube.ru>;tag=e385ad108f5b4309bab3920a9fd91355.045b
Call-ID: 3832e75741d8f5245078e4d12248bd2e@sip.comtube.ru
CSeq: 104 INVITE
Proxy-Authenticate: Digest realm="sip.comtube.ru", nonce="4ff39cd700011ea4622e3355e68b5ee6068d11851c335447", stale=true
Server: Comtube SIP Proxy
Content-Length: 0
<------------->
--- (9 headers 0 lines) ---
<--- SIP read from UDP:62.117.120.98:5060 --->
SIP/2.0 180 Ringing
Content-Type:application/sdp
Allow:INVITE,ACK,OPTIONS,BYE,CANCEL,REGISTER,INFO,UPDATE,PRACK,REFER,SUBSCRIBE,NOTIFY
Date:Sun, 02 Jan 2000 08:15:01 GMT
From:sip:295039@sip.comtube.ru;tag=as4785c625
To:<sip:79242245055@sip.comtube.ru>;tag=EE1D3030353039320011E4D9
Call-ID:3832e75741d8f5245078e4d12248bd2e@sip.comtube.ru
CSeq:103 INVITE
Record-Route:<sip:62.117.120.98;lr;ftag=as4785c625;did=11e.dee8ca51>
Server:TB005092/2.1
Via:SIP/2.0/UDP 193.169.37.103:5060;received=193.169.37.103;branch=z9hG4bK2d030531;rport=5060
Content-Length: 157
<--- SIP read from UDP:62.117.120.98:5060 --->
SIP/2.0 183 Session Progress
Content-Type:application/sdp
Allow:INVITE,ACK,OPTIONS,BYE,CANCEL,REGISTER,INFO,UPDATE,PRACK,REFER,SUBSCRIBE,NOTIFY
Date:Sun, 02 Jan 2000 08:15:05 GMT
From:sip:295039@sip.comtube.ru;tag=as4785c625
To:<sip:79242245055@sip.comtube.ru>;tag=EE1D3030353039320011E4D9
Call-ID:3832e75741d8f5245078e4d12248bd2e@sip.comtube.ru
CSeq:103 INVITE
Record-Route:<sip:62.117.120.98;lr;ftag=as4785c625;did=11e.dee8ca51>
Server:TB005092/2.1
Via:SIP/2.0/UDP 193.169.37.103:5060;received=193.169.37.103;branch=z9hG4bK2d030531;rport=5060
Content-Length: 157
<--- SIP read from UDP:62.117.120.98:5060 --->
SIP/2.0 200 OK
Content-Type:application/sdp
Contact:sip:79242245055@192.168.3.10:5061;nat=yes
Allow:INVITE,ACK,OPTIONS,BYE,CANCEL,REGISTER,INFO,UPDATE,PRACK,REFER,SUBSCRIBE,NOTIFY
Supported:100rel
Accept:application/sdp
From:sip:295039@sip.comtube.ru;tag=as4785c625
To:<sip:79242245055@sip.comtube.ru>;tag=EE1D3030353039320011E4D9
Call-ID:3832e75741d8f5245078e4d12248bd2e@sip.comtube.ru
CSeq:103 INVITE
Record-Route:<sip:62.117.120.98;lr;ftag=as4785c625;did=11e.dee8ca51>
Server:TB005092/2.1
Via:SIP/2.0/UDP 193.169.37.103:5060;received=193.169.37.103;branch=z9hG4bK2d030531;rport=5060
Content-Length: 157
Собственно, получается что сначала sip proxy отбивает с 403 ошибкой, но при этом звонок идет дальше. Номер, который я набирал, звонит, не смотря на то, что с моей стороны звонок уже завершен.
Регистрация работает.
- Код: Выделить всё
comtube-out/295039 62.117.120.98 N 5060 OK (119 ms)
Вопрос - что с этим делать?
P.S. Транки с другими операторами работают.
